对asterisk的一些研究

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发布于 2021-05-19 11:21:54

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发布于 2021-05-19 11:21:54

文章被收录于专栏：全栈程序员必看

这段时间一直在研究asterisk，是基于《Asterisk™ The Future of Telephony》这本书展开的，涉及asterisk的安装，调试，SIP，IAX，以及一些基本的配置等，这里对测试的脚本进行留存

因为我们用的asterisk大部分都装了 freepbx等，配置文件看起来超复杂，找不到重点，这里的保留最原始的。。

配置SIP分机用的，这个文件其实可以超简单的。。

sip.conf

[general] register => tontone:123456@192.168.0.105/asaka

[asaka] type=friend host=192.168.0.105 context=asaka_incoming secert=123456

[1000] type=friend host=dynamic context=from-internal

[2000] type=friend host=dynamic context=from-internal ;requirecalltoken=no

配置IAX用的。。

iax.conf

[general] autokill=yes

[tontone] type=friend secret=123456 host=dynamic context=incoming_tontone trunk=yes ;requirecalltoken=no

[zoiper] type=friend host=dynamic context=from-internal

配置dahdi

chan_dahdi.conf

;# Flash Operator Panel will parse this file for dahdi trunk buttons ;# AMPLABEL will be used for the display labels on the buttons

;# %c Dahdi Channel number ;# %n Line number ;# %N Line number, but restart counter ;# Example: ;# ;AMPLABEL:Channel %c – Button %n

;# For Dahdi/* buttons use the following ;# (where x=number of buttons to dislpay) ;# ;AMPWILDCARDLABEL(x):MyLabel

[channels] language=en

; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf #include dahdi-channels.conf

; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c – Button %n ;context=from-pstn ;signalling=fxs_ks ;faxdetect=incoming ;usecallerid=yes ;echocancel=yes ;echocancelwhenbridged=no ;echotraining=800 ;group=0

;channel=1-2

usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes immediate=no callerid=4001 ; define channels

context=from-internal ; Uses the [internal] chntext in extensions.conf signalling=fxo_ks ; Uses FXO signalling for an FXS channel channel => 1 ; Telephone attached to port 1

context=from-pstn ; Incoming calls go to [incoming] in extensions.conf signalling=fxs_ks ; Use FXS signalling for an FXO channel channel => 2 ; PSTN attached to port 2

dahdi-channels.conf

; Autogenerated by /usr/sbin/dahdi_genconf on Tue Jun 22 16:59:35 2010 — do not hand edit ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings ;

; Span 1: WCTDM/4 “Wildcard S400P Prototype Board 5” (MASTER) ;;; line=”1 WCTDM/4/0″ signalling=fxo_ls callerid=”Channel 1″ <4001> mailbox=4001 group=5 context=from-internal channel => 1 callerid= mailbox= group= context=default

;;; line=”2 WCTDM/4/1″ signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 2 callerid= group= context=default

配置拔号方案 extensions.conf

[globals] OUTBOUNDTRUNK=DAHDI/2 TELE=DAHDI/1 ZOIPER=IAX2/zoiper

[general] autofallthrough=yes

[default]exten => s,1,Verbose(1|Unrouted call handler)exten => s,n,Answer()exten => s,n,Wait(1)exten => s,n,Playback(tt-weasels);exten => s,n,Dial(sip/1000,20);exten => s,n,Record(/var/spool/asterisk/monitor/asterisk-

[incoming_tontone] include => from-internal

exten => _105XXX.,1,Verbose(1|exten is 105XXX) exten => _105XXX.,n,NoOp() exten => _105XXX.,n,Dial(IAX2/tontone/${EXTEN:3},20) exten => _105XXX.,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _105XXX.,n,Hangup()

[asaka_incoming] exten => _135XXX.,1,Verbose(1|exten is 1055XXXX) exten => _135XXX.,n,NoOp() exten => _135XXX.,n,Dial(SIP/asaka/${EXTEN:3},20) exten => _135XXX.,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _135XXX.,n,Hangup()

include => incoming_tontone include => internal include => call-out

[from-internal] include => internal include => incoming_tontone include => asaka_incoming include => call-out include => test-waitexten

[internal] exten => _XXXX,1,Verbose(1|Unrouted call handler) exten => _XXXX,n,Answer() exten => _XXXX,n,Wait(1.2) exten => _XXXX,n,Dial(SIP/${EXTEN},20) ;exten => _XXXX,n,VoiceMail(2000@default,u) exten => _XXXX,n,hangup()

exten => 8,1,Directory(default,incoming,f) exten => 9,1,Directory(default,incoming)

exten => 456,1,Set(DB(test/count)=1)exten => 456,n,Set(COUNT=

exten => 500,1,Macro(voicemail,SIP/2000)

exten => 600,1,MeetMeCount(600,CONFCOUNT)exten => 600,n,GotoIf(

exten => 601,1,Playback(conf-thereare) exten => 601,n,MeetMeCount(600) exten => 601,n,Playback(conf-peopleinconf)

exten => 777,1,Macro(mysql,15921256331)

[test-waitexten] exten => 123,1,Answer() exten => 123,n,Background(enter-ext-of-person) exten => 123,n,WaitExten()

exten => 2,1,playback(digits/2) exten => 2,n,Goto(123,1)

exten => 3,1,playback(digits/3) exten => 3,n,Goto(123,1)

exten => i,1,playback(pbx-invalid) exten => i,n,Goto(123,1)

exten => t,1,playback(vm-goodbye) exten => t,n,hangup()

[macro-voicemail-a]exten => s,1,Dial({DIALSTATUS}” = “BUSY”]?busy:unavail)exten => s,n(unavail),Voicemail(

[macro-voicemail]exten => s,1,Dial({MACRO_EXTEN},u)exten => s-NOANSWER,n,Goto(incoming,s,1)exten => s-BUSY,1,Voicemail(

[macro-mysql]exten => s,1,Set(NUM_tmp={NUM_tmp:0:1}=1]?judge)exten => s,n,Set(DIAL_NUMBER={Num_tmp:0:7})exten => s,n,MYSQL(Connect connid localhost freepbx fpbx test)exten => s,n,MYSQL(Query resultid {ExtenPre})exten => s,n,MYSQL(Fetch fechid {fechid})exten => s,n,Set(DIAL_NUMBER=[{fechid}=0]?0{NUM_tmp}:{NUM_tmp}))exten => s,n,MYSQL(Clear {resultid})exten => s,n,MYSQL(Disconnect

[call-out]exten => _XXX.,1,answer()exten => _XXX.,n,wait(1)exten => _XXX.,n,Monitor(wav,asterisk-monitor-

[from-pstn] ;exten => s,1,Zapateller(nocallerid) ;exten => s,n,Playback(enter-ext-of-person)

exten => s,1,answer()exten => s,n,wait(1.5);exten => s,n,Monitor(wav,asterisk-monitor-

[dial-tele]exten => s,1,Monitor(wav,asterisk-monitor-{ZOIPER},20);exten => s,n,Verbose(1|test tele [“{DIALSTATUS}”=”CANCEL”]?cancel:)exten => s,n(cancel),Hangup()

配置voicemail

voicemail.conf

[general] #include vm_general.inc #include vm_email.inc [default] 2000 => 1234,Aaron,evane1890@gmail.com,chen_jiang_tao@hotmail.com 500 => 1234,Aaron2,evane1890@gmail.com,chen_jiang_tao@hotmail.com